Use suffixed {uint,int}{8,16,32,64}_t types.

webrtc/base/asyncpacketsocket.h

 /*
  *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 10 matching lines @@
 // This structure holds the info needed to update the packet send time header
 // extension, including the information needed to update the authentication tag
 // after changing the value.
 struct PacketTimeUpdateParams {
   PacketTimeUpdateParams();
   ~PacketTimeUpdateParams();
 
   int rtp_sendtime_extension_id;    // extension header id present in packet.
   std::vector<char> srtp_auth_key;  // Authentication key.
   int srtp_auth_tag_len;            // Authentication tag length.
-  int64 srtp_packet_index;          // Required for Rtp Packet authentication.
+  int64_t srtp_packet_index;        // Required for Rtp Packet authentication.
 };
 
 // This structure holds meta information for the packet which is about to send
 // over network.
 struct PacketOptions {
   PacketOptions() : dscp(DSCP_NO_CHANGE) {}
   explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp) {}
 
   DiffServCodePoint dscp;
   PacketTimeUpdateParams packet_time_params;
 };
 
 // This structure will have the information about when packet is actually
 // received by socket.
 struct PacketTime {
   PacketTime() : timestamp(-1), not_before(-1) {}
-  PacketTime(int64 timestamp, int64 not_before)
-      : timestamp(timestamp), not_before(not_before) {
-  }
+  PacketTime(int64_t timestamp, int64_t not_before)
+      : timestamp(timestamp), not_before(not_before) {}
 
-  int64 timestamp;  // Receive time after socket delivers the data.
-  int64 not_before; // Earliest possible time the data could have arrived,
-                    // indicating the potential error in the |timestamp| value,
-                    // in case the system, is busy. For example, the time of
-                    // the last select() call.
-                    // If unknown, this value will be set to zero.
+  int64_t timestamp;   // Receive time after socket delivers the data.
+
+  // Earliest possible time the data could have arrived, indicating the
+  // potential error in the |timestamp| value, in case the system, is busy. For
+  // example, the time of the last select() call.
+  // If unknown, this value will be set to zero.
+  int64_t not_before;
 };
 
-inline PacketTime CreatePacketTime(int64 not_before) {
+inline PacketTime CreatePacketTime(int64_t not_before) {
   return PacketTime(TimeMicros(), not_before);
 }
 
 // Provides the ability to receive packets asynchronously. Sends are not
 // buffered since it is acceptable to drop packets under high load.
 class AsyncPacketSocket : public sigslot::has_slots<> {
  public:
   enum State {
     STATE_CLOSED,
     STATE_BINDING,
@@ skipping 58 matching lines @@
   // Used only for listening TCP sockets.
   sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AsyncPacketSocket);
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_ASYNCPACKETSOCKET_H_